Network Working Group Magnus Westerlund
INTERNET-DRAFT Ericsson
Category: Standards Track February 24, 2003
Expires: August 2003
A Transport Independent Bandwidth Modifier for the Session
Description Protocol (SDP).
Status of this memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that other
groups may also distribute working documents as Internet-Drafts.
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This document is an individual submission to the IETF. Comments
should be directed to the authors.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
The existing Session Description Protocol (SDP) bandwidth modifiers
and their values include the bandwidth needed also for the transport
and IP layers. When using SDP in protocols like Session Announcement
Protocol (SAP), Session Initiation Protocol (SIP) and Real-Time
Streaming Protocol (RTSP) and the involved hosts reside in networks
running different IP versions, the interpretation of what type of
lower layers that is included is not clear. This documents defines a
bandwidth modifier that does not include transport overhead, instead
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an additional packet rate attribute is defined. The transport
independent bit-rate value together with the maximum packet rate can
then be used to calculate the real bit-rate over the actually used
transport.
TABLE OF CONTENTS
1. Definitions.........................................................3
1.1. Glossary.......................................................3
1.2. Terminology....................................................3
2. Introduction........................................................3
2.1. The Bandwidth Attribute........................................3
2.1.1. Conference Total..........................................3
2.1.2. Application Specific Maximum..............................3
2.1.3. RTCP Report bandwidth.....................................4
2.2. IPv6 and IPv4..................................................4
3. The Bandwidth Signaling Problems....................................5
3.1. What IP version is used........................................5
3.2. Converting bandwidth values....................................6
3.3. Header Compression.............................................6
3.4. RTCP problems..................................................7
3.5. Future development.............................................7
4. Problem Scope.......................................................7
5. Requirements........................................................8
6. A Solution..........................................................8
6.1. Introduction...................................................8
6.2. The TIAS bandwidth modifier....................................8
6.2.1. Usage.....................................................8
6.2.2. Definition................................................9
6.2.3. Usage Rules..............................................10
6.3. Packet Rate parameters........................................10
6.4. Converting to Transport Dependent values......................11
6.5. Deriving RTCP bandwidth.......................................11
6.5.1. Motivation to being an acceptable solution...............11
6.6. ABNF definitions..............................................12
7. Protocol Interaction...............................................12
7.1. RTSP..........................................................12
7.2. SIP...........................................................12
7.3. SAP...........................................................13
8. Security Consideration.............................................13
9. IANA Consideration.................................................13
10. Acknowledgments...................................................13
11. Author's Addresses................................................14
12. References........................................................14
12.1. Normative references.........................................14
12.2. Informative References.......................................14
13. IPR Notice........................................................15
14. Copyright Notice..................................................16
15. Changes...........................................................16
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1. Definitions
1.1. Glossary
RTSP - Real-Time Streaming Protocol, see [8].
SDP - Session Description Protocol, see [6].
SAP - Session Announcement Protocol, see [5].
SIP - Session Initiation Protocol, see [6].
TIAS - Transport Independent Application Specific maximum, a
bandwidth modifier.
1.2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [3].
2. Introduction
Today the Session Description Protocol (SDP) [1] is used in several
types of applications. The original application is session
information and configuration for multicast sessions announced with
Session Announcement Protocol (SAP) [5]. SDP is also a vital
component in media negotiation for the Session Initiation Protocol
(SIP) [6] by using the offer answer model [7]. The Real-Time
Streaming Protocol (RTSP) [8] also makes use of SDP to declare what
media and codec(s) a multi-media presentation consist of to the
client.
2.1. The Bandwidth Attribute
In SDP [1] there exist a bandwidth attribute, which has a modifier
used to specify what type of bit-rate the value refers to. The
attribute has the following form:
b=:
Today there are four modifiers defined which are used for different
purposes.
2.1.1. Conference Total
The Conference Total is indicated by giving the modifier "CT". The
meaning of Conference total is to give a maximum bandwidth that a
conference session will use. Its purpose is to decide if this session
can co-exist with any other sessions. Defined in RFC 2327 [1].
2.1.2. Application Specific Maximum
The Application Specific maximum bandwidth is indicated by the
modifier "AS". The interpretation of this attribute is depending on
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the application's notion of maximum bandwidth. For an RTP application
this attribute is the RTP session bandwidth as defined in RFC 1889
[4]. The session bandwidth includes the bandwidth that the RTP data
traffic will result in, including the lower layers down to IP layer.
So the bandwidth is in most cases calculated over RTP payload, RTP
header, UDP and IP. Defined in RFC 2327 [1].
2.1.3. RTCP Report bandwidth
Today there is a draft [9], currently in the RFC editors queue to
become a Proposed Standard, which defines two new bandwidth
modifiers. These modifiers "RS" and "RR", define the amount of
bandwidth that is assigned for RTCP reports by active data senders
respectively RTCP reports by receivers only.
2.2. IPv6 and IPv4
Today there are two IP versions 4 [15] and 6 [14] used in parallel on
the Internet. This creates problems and there exist a number of
possible transition mechanisms.
------------------ ----------------------
| IPv4 domain | | IPv6 Domain |
| | ---------| | |
| ---------- |-| NAT-PT |-| ---------- |
| |Server A| | ---------| | |Client B| |
| ---------- | | ---------- |
------------------ ----------------------
Figure 1. Translation between IPv6 and IPv4 addresses.
- To achieve connectivity between an IPv4 only host and an IPv6
only host, translation is necessary. This translator can be for
example Network Address Translation - Protocol Translation (NAT-
PT) [13], see Figure 1. However to get connectivity for large
number of protocols, Application Level Gateway (ALG) functionality
is also required at the node. To be able to locate hosts through
the translation node DNS ALG must be supported.
- IPv6 nodes belonging to different domains running IPv6, but
lacking IPv6 connectivity between them solves this by tunneling
over the IPv4 net, see Figure 2. Basically the IPv6 packets are
put as payload in IPv4 packets between the tunneling end-points at
the edge of each IPv6 domain.
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--------------- --------------- ---------------
| IPv6 domain | | IPv4 domain | | IPv6 Domain |
| | |-------------| | |
| ---------- |--||Tunnel ||--| ---------- |
| |Server A| | |-------------| | |Client B| |
| ---------- | | | | ---------- |
--------------- --------------- --------------|
Figure 2. Tunneling through a IPv4 domain
IPv4 has minimal header size of 20 bytes. While the fixed part of the
IPv6 header is 40 bytes.
The difference in header sizes, results in that the bit-rate required
for a certain IP layer is different. How big the difference is,
depends on the packet rate and size difference of each packet.
3. The Bandwidth Signaling Problems
When an application wants to use SDP to signal the bandwidth required
for this application some problems becomes evident depending on the
transport layers.
3.1. What IP version is used
If one signals the bandwidth in SDP, with for example "b=AS:" for an
RTP based application, one cannot know if the overhead is calculated
for IPv4 or IPv6. An indication to which protocol has been used when
calculating the bandwidth values is given by the "c=" connection data
line. This line contains either a multicast group address or a
unicast address of the data source or sink. The "c=" lines address
type may be assumed to be of the same type as the one used in the
bandwidth calculation. There seems to exist no specification pointing
this out.
In cases of SDP transported by RTSP this is even less clear. The
normal usage for a unicast on-demand streaming session is to set the
connection data address to a null address. This null address does
have an address type, which could be used as an indication. However
this is also not clarified anywhere.
Figure 1, illustrates a connection scenario between a streaming
server A and a client B over a translator here designated as a NAT-
PT. When B receives the SDP from A over RTSP it will be very
difficult for B to know what the bandwidth values in the SDP
represent. The following possibilities exist:
1. The SDP is unchanged and "c=" null address is of type IPv4. The
bandwidth value represents the bandwidth needed in an IPv4
network.
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2. The SDP has been changed by the ALG. The "c=" address is changed
to IPv6 type. The bandwidth value is unchanged.
3. The SDP is changed and both "c=" address type and bandwidth value
is changed. Unfortunately, this can seldom be done, see 3.2.
In case 1 the client can understand that the server is located in an
IPv4 network and that it uses IPv4 overhead when calculating the
bandwidth value. The client can almost never convert the bandwidth
value, see section 3.2.
In case 2 the client does not know that the server is in an IPv4
network and that the bandwidth value is not calculated with IPv6
overhead. In cases where a client reserve bandwidth for this flow,
too little will be reserved, potentially resulting in bad Quality of
Service (QoS).
In case 3 everything works correctly. However this case will be very
rare. If one tries to convert the bandwidth value without further
information about the packet rate significant errors may be
introduced into the value.
3.2. Converting bandwidth values
If one would like to convert a bandwidth value calculated using IPv4
overhead to IPv6 overhead the packet rate is required. The new
bandwidth value for IPv6 is normally "IPv4 bandwidth" + "packet rate"
* 20 bytes. Where 20 bytes is the usual difference between IPv6 and
IPv4 headers. The overhead difference may be other in cases when IPv4
options [15] or IPv6 extension headers [14] are used.
As converting requires the packet rate of the stream, this is not
possible in the general case. Many codecs has many possible packet
rates. Therefore some extra information in the SDP will be required.
The "a=ptime:" parameter may be a possible candidate. However this
parameter is normally only used for audio codecs. Also its definition
[1] is that it is only a recommendation, which the sender may
disregard from. A better parameter is needed.
3.3. Header Compression
Another mechanism that alters the actual overhead over links is
header compression. Header compression uses the fact that most
network protocol headers have either static or predictable values in
their fields within a packet stream. Compression is normally only
done on per hop basis, i.e. on a single link. The normal reason for
doing header compression is that the link has fairly limited
bandwidth and significant gain in throughput is achieved.
There exist a couple of different header compression standard. For
compressing IP headers only, there exist RFC 2507 [10]. For
compressing packets with IP/UDP/RTP headers, CRTP [11] was created at
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the same time. More recently the Robust Header Compression (ROHC)
working group has been developing a framework and profiles [12] for
compressing certain combinations of protocols, like IP/UDP, and
IP/UDP/RTP.
When using header compression the actual used overhead will be less
deterministic, but in most cases an average overhead can be
determined for a certain application. If a network node knows that
some type of header compression is employed this can taken into
consideration. To be able to do this with any accuracy the
application and packet rate is needed.
3.4. RTCP problems
When RTCP is used between host in IPv4 and IPv6 networks over an NAT-
PT, similar problems exist. The RTCP traffic going from the IPv4
domain will result in a higher RTCP bit-rate than intended in the
IPv6 domain due to the larger headers. This may result in up to 25%
increase in required bandwidth for the RTCP traffic. The largest
increase will be for small RTCP packets when the number of IPv4 hosts
is much larger than the number of IPv6 hosts. Fortunately as RTCP has
a limited bandwidth compared to RTP it will only result in a maximum
of 1.75% increase of the total session bandwidth when RTCP bandwidth
is 5% of RTP bandwidth. The RTCP randomization may easily result in
short term effects of the same magnitude. The increase in bandwidth
will in most cases be less.
At the same time this results in unfairness in the reporting between
an IPv4 and IPv6 node. The IPv6 node may report in the worst case
with 25% longer intervals.
These problems have been considered insignificant enough to not be
worth any complex solutions. Therefore only a simple algorithm for
deriving RTCP bandwidth is defined.
3.5. Future development
Today there is work in IETF to design a new datagram transport
protocol, which will be suitable to use for real-time media. This
protocol is called the Datagram Congestion Control Protocol (DCCP).
This protocol will most probably have another header size than UDP,
which is mostly used today. This results in even further numbers of
possible transport combinations to calculate overhead for.
4. Problem Scope
The problems described in chapter 3 does effect all the protocols
that uses SDP to signal bandwidth parameters including transport
level bit-rates.
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In the MMUSIC WG there is work on a replacement of SDP called SDP-NG.
That work is RECOMMENDED to consider the problems outlined in this
draft when designing solutions for specifying bandwidth in SDP-NG.
5. Requirements
A solution to the problems outlined in this draft should meet the
following requirements:
- The bandwidth value SHALL be given in a way so that it can be
calculated for all possible combinations of transport overhead.
6. A Solution
6.1. Introduction
This chapter describes a solution for the problems outlined in this
document for the Application Specific (AS) bandwidth modifier.
The CT is a session level modifier and cannot easily be dealt with.
To address the problems with different overhead the CT value is
RECOMMENDED to be calculated using reasonable worst case overhead.
The RR and RS modifiers [9] will be used as defined and includes
transport overhead. The small unfairness between hosts is deemed
acceptable.
6.2. The TIAS bandwidth modifier
6.2.1. Usage
A new bandwidth modifier is defined to be used for the following
purposes:
- Resource reservation. A single bit-rate can be enough to use for
resource reservation. Some characteristics can be derived from the
stream, codec type, etc. In cases where more information is
needed, then another SDP parameter will be required.
- Maximum media codec rate. With the definition below of "TIAS" the
given bit-rate will mostly be from the media codec. Therefore it
gives a good indication on the maximum codec bit-rate required to
be supported by the decoder.
- Communication bit-rate required for the stream. The "TIAS" value
together with "maxprate" can be used to determine the maximum
communication bit-rate the stream will require. By adding all
maximum bit-rates from the streams in a session together, a
receiver can determine if its communication resources are
sufficient to handle the stream. For example a modem user can
determine if the session fits his modems capabilities and the
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established connection.
- Determine the RTP session bandwidth and derive the RTCP
bandwidth. The derived transport dependent attribute will be the
RTP session bandwidth in case of RTP based transport. The TIAS
value can also be used to determine the RTCP bandwidth to use when
using implicit allocation. RTP [4] defines that if not explicitly
stated, additional bandwidth shall be used by RTCP equal to 5% of
the RTP session bandwidth. The RTCP bandwidth can be explicitly
allocated by using the RR and RS modifiers defined in [9].
6.2.2. Definition
A new session and media level bandwidth modifier is defined:
b=TIAS: ; see section 6.6 for ABNF definition.
The Transport Independent Application Specific Maximum (TIAS)
bandwidth modifier has an integer bit-rate value in bits per second.
A fractional bandwidth value SHALL always be rounded up to the next
integer. The bandwidth value is the maximum needed by the application
(session level) or media stream (media level) without counting IP and
other transport layers like TCP or UDP. On session level the TIAS
value is simply the sum of all media stream's TIAS values. For RTP
based media streams, TIAS on media level can be used to derive the
RTP "session bandwidth" as defined in section 6.2 of [4]. However in
the context of RTP transport the TIAS value is defined as:
Only the RTP payload as defined in [4], SHALL be used in the
calculation of the bit-rate, i.e. the lower layers (IP/UDP) and
RTP headers including RTP header, RTP header extensions, CSRC list
and other RTP profile specific fields are excluded. Note that the
RTP payload includes both payload formats headers and its data
parts. This may allow one to use the same value for both RTP based
media transport as non-RTP transport and stored media.
Note 1: The usage of bps is not in accordance with RFC 2327 [1]. This
change has no implications on the parser, only the interpreter of the
value must be aware. The change is done to allow for better
resolution, and has also been used for the RR and RS bandwidth
modifiers see [9].
Note 2: RTCP bandwidth is not included in the bandwidth value. In
applications using RTCP, the bandwidth used by RTCP is either 5% of
the RTP session bandwidth including lower layers or as specified by
the RR and RS modifiers [9]. A definition on how to derive the RTCP
bit-rate when using TIAS is present in chapter 6.5.
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6.2.3. Usage Rules
"TIAS" is primarily intended to be used at media level. The TIAS
bandwidth attribute MAY be present on session level in the SDP.
However, if present at session level it SHOULD be present also on
media level. "TIAS" SHALL NOT be present on session level, unless the
same transport is used for all media streams. The session level value
if present MUST be the sum of all media levels.
To allow for backwards compatibility towards users of SDP that does
not implement "TIAS", it is RECOMMENDED to also include the "AS"
modifier when using "TIAS". The presence of any value, even with
problems is better than none. However an SDP user understanding TIAS
when present SHOULD ignore the "AS" value and use TIAS instead.
When using TIAS for an RTP transported stream(s) the "maxprate"
attribute SHALL be included at the corresponding SDP level.
6.3. Packet Rate parameters
To be able to calculate the bandwidth value including the actually
used lower layers, a packet rate attribute is also defined.
The session and media level maximum packet rate attribute is defined
as:
a=maxprate: ; see section 6.6 for ABNF definition.
The is a floating-point value for the streams maximum
packet rate. If the number of packets is variable the given value
SHALL be the maximum the application, can produce in case of live
stream, or for on-demand streams, have produced. The packet rate is
calculated by adding together the number of packets sent within a 1
second long window. The maxprate is the largest value produced when
the window is slide over the entire media stream. In cases that this
can't be calculated, i.e. for example a live stream, a estimated
value of the maximum rate the codec can produce for the given
configuration and content SHALL be used.
A session level value MUST be the sum of all media level packet
rates. The session level value MAY only be given if all media streams
uses the same transport. If that is not the case, the "maxprate"
attribute MUST NOT be present at session level. If given at session
level it SHOULD also be given at media level.
"maxprate" SHOULD be included for all transports where a packet rate
can be derived and TIAS is included.
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6.4. Converting to Transport Dependent values
When converting the transport independent bit-rate value (bw-value)
into a transport dependent value including the lower layers the
following MUST be done:
1. Determine which lower layers that will be used and calculate the
sum of the sizes of the headers in bits (h-size). In cases of
variable header sizes the average size SHALL be used. For RTP
transported media the lower layers SHALL include the RTP header
with used header extensions, CSRC list, and profile specific
extensions.
2. Retrieve the maximum packet rate from the SDP (prate = maxprate).
3. Calculate the transport overhead by multiplying the header sizes
with the packet rate (t-over = h-size * prate).
4. Round the transport overhead up to nearest integer in bits(t-over
= CEIL(t-over)).
5. Add the transport overhead to the transport independent bit-rate
value (bit-rate = bw-value + t-over)
When the above calculation is performed using the "maxprate" the bit-
rate value will be the absolute maximum the media stream may use over
the transport used in the calculations.
6.5. Deriving RTCP bandwidth
This chapter does not solve the fairness and possible bit-rate change
introduced by IPv4 to IPv6 translation. These differences are
considered small enough and known solutions introduce code changes to
the RTP/RTCP implementation. This chapter gives only a consistent way
of calculating the bit-rate to assign to RTCP if not explicitly
given.
First the transport dependent RTP session bit-rate is calculated, in
accordance with chapter 6.4, using the actual transport layers used
at this end point. The RTCP bit-rate is then derived as usual based
on the RTP session bandwidth, i.e. normally equal to 5% of the
calculated value.
6.5.1. Motivation to being an acceptable solution.
Giving the exact same bit-rate value to both the IPv4 and IPv6 host
will result in that the IPv4 host has a higher RTCP sending rate. For
a 100 bytes RTCP packet (including UDP/IPv4) the IPv4 sender has
approximate 20 % higher rate. This rate falls with larger RTCP
packets. For example, 300 bytes packets will only give the IPv4 host
a 7% higher reporting rate.
The above rule for deriving RTCP bandwidth, gives the same behavior
as fixed assignment when the RTP session has traffic parameters
giving a large TIAS/maxprate ratio. The two hosts will be fair when
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the TIAS/maxprate ratio is approximate 40 (bytes/packet) given 100
bytes RTCP packets. For a TIAS/maxprate ratio of 5 bytes/packet the
IPv6 host will be allowed to send approximately 15-20 % more RTCP
packets. The larger the RTCP packets become the more it will favor
the IPv6 host in sending rate.
The conclusions is that the within the normal useful combination of
transport independent bit-rates and packet rates the difference in
fairness between a host on different IP versions with different
overhead is acceptable. For the 20 bytes difference in overhead
between IPv4 and IPv6 headers the actually used RTCP bandwidth in a
unicast connection case will not be larger than approximately 1% of
the total session bandwidth.
6.6. ABNF definitions
This chapter defines in ABNF from RFC 2234 [2] the bandwidth modifier
and the packet rate attribute.
The bandwidth modifier:
TIAS-bandwidth-def = "b" "=" "TIAS" ":" bandwidth-value
bandwidth-value = 1*DIGIT
The maximum packet rate attribute:
max-p-rate-def = "a" "=" "maxprate" ":" packet-rate CRLF
packet-rate = 1*DIGIT ["." 1*DIGIT]
7. Protocol Interaction
7.1. RTSP
The "TIAS", and "maxprate" can today be used with RTSP. To be able to
calculate the transport dependent bandwidth, some of the transport
header parameters will be required. There should be no problems for a
client to calculate the required bandwidth(s) prior to a RTSP SETUP.
The reason is that a client supports a limited number of transport
setups. The one actually offered a server in a SETUP request will be
dependent on the contents of the SDP description. The "m=" line(s)
will signal to the client the desired transport profile(s).
7.2. SIP
The usage of "TIAS" together with "maxprate" should not be different
from the handling of the "AS" modifier currently in use. The needed
transport parameters will available in the transport field in the
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"m=" line. The address class can be determined from the "c=" field
and the clients connectivity.
7.3. SAP
In the case of SAP all available information to calculate the
transport dependent bit-rate should be present in the SDP. The "c="
information gives the used address family for the multicast. The
transport layer, e.g. RTP/UDP, for each media is evident in the media
line ("m=") and its transport field.
8. Security Consideration
The bandwidth value that is supplied by the parameters defined here
can, if not protected, be altered. By altering the bandwidth value
one can fool a receiver to reserve either more or less bandwidth than
actually needed. Reserving too much may result in unwanted expenses
on behalf of user and also blocking of resources that other parties
could have used. If to little bandwidth is reserved the receiving
users quality MAY be effected. Trusting a to large TIAS value may
also result in that the receiver will turn down the session due to
insufficient communication and decoding resources.
Due to these security risks it is STRONGLY RECOMMENDED that the SDP
is authenticated so no tampering can be performed. It is also
RECOMMENDED that any receiver of the SDP performs an analysis of the
received bandwidth values so that they are reasonable and is what can
be expected for the application. For example, a single channel AMR
encoded voice stream claiming to use 1000 kbps is not reasonable.
9. IANA Consideration
This document register one new SDP session and media level attribute
"maxprate", see section 6.3.
A new SDP [1] bandwidth modifier (bwtype) "TIAS" is also registered
in accordance with the rules requiring a standard tracks RFC. The
modifier is defined in section 6.2.
10. Acknowledgments
The author would like to thank Gonzalo Camarillo and Hesham Soliman
for their work reviewing this document.
The author would also like to thank all persons on the MMUSIC working
group's mailing list that has commented on this specification.
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11. Author's Addresses
Magnus Westerlund Tel: +46 8 4048287
Ericsson Research Email: Magnus.Westerlund@ericsson.com
Ericsson AB
Torshamnsgatan 23
SE-164 80 Stockholm, SWEDEN
12. References
12.1. Normative references
[1] M. Handley, V. Jacobson, "Session Description Protocol (SDP)",
IETF RFC 2327, April 1998.
[2] D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov.
1997.
[3] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", RFC 2119, March 1997.
[4] H. Schulzrinne, et. al., "RTP: A Transport Protocol for Real-
Time Applications", IETF RFC 1889, January 1996.
12.2. Informative References
[5] M. Handley et al., "Session Announcement Protocol", IETF RFC
2974, October 2000.
[6] J. Rosenberg, et. al., "SIP: Session Initiation Protocol", IETF
RFC 3261, June 2002.
[7] J. Rosenberg, H. Schulzrine, "An Offer/Answer Model with Session
Description Protocol (SDP)", IETF RFC 3164, June 2002.
[8] H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
IETF RFC 2326, April 1998.
[9] S. Casner, "SDP Bandwidth Modifiers for RTCP Bandwidth", IETF WG
draft, draft-ietf-avt-rtcp-bw-05.txt, November 2001, Work in
progress
[10] M. Degermark, B. Nordgren, S. Pink, "IP Header Compression",
IETF RFC 2507, February 1999.
[11] S. Casner, V. Jacobson, "Compressing IP/UDP/RTP Headers for Low-
Speed Serial Links", IETF RFC 2508, February 1999.
[12] C. Bormann, et. al., "RObust Header Compression (ROHC):
Framework and four profiles", IETF RFC 3095, July 2001.
[13] Tsirtsis, G. and Srisuresh, P., "Network Address Translation -
Protocol Translation (NAT-PT)", RFC 2766, Internet Engineering
Task Force, February 2000.
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[14] S. Deering and R. Hinden, "Internet Protocol, Version 6 (IPv6)
Specification", RFC 2460, Internet Engineering Task Force,
December 1998.
[15] J. Postel, "internet protocol", RFC 791, Internet Engineering
Task Force, September 1981.
13. IPR Notice
The IETF takes no position regarding the validity or scope of any
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The IETF invites any interested party to bring to its attention any
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this standard. Please address the information to the IETF Executive
Director.
Westerlund Standards Track [Page 15]
INTERNET-DRAFT Bandwidth modifier for SDP Feb. 24, 2003
14. Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and
furnished to others, and derivative works that comment on or
otherwise explain it or assist in its implementation may be
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not be modified in any way, such as by removing the copyright
notice or references to the Internet Society or other Internet
organizations, except as needed for the purpose of developing
Internet standards in which case the procedures for copyrights
defined in the Internet Standards process must be followed, or
as required to translate it into languages other than English.
The limited permissions granted above are perpetual and will
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This document and the information contained herein is provided
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ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR
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15. Changes
[Note to RFC Editor: Remove this section when publishing]
The following changes has been done to this version compared to
draft-ietf-mmusic-sdp-bwparam-00.txt:
- Clarified definition of TIAS value for RTP.
- A few minor wording changes.
This Internet-Draft expires in August 2003.
Westerlund Standards Track [Page 16]